1. Field of the Invention
This invention relates to a method of and architecture for obtaining an instantaneous unambiguous bandwidth beyond the analog to digital (A/D) converter sample rate in a digital receiver using current technology A/D converters and, more specifically, to obtaining such a bandwidth by determining the true frequency of an undersampled input signal after it has been mapped into the sample frequency bandwidth.
2. Brief Description of the Prior Art
Traditionally, incoming analog signals which are to be converted to digital signals are sampled at a rate of at least twice the receiver bandwidth in order to satisfy the Nyquist criteria. To achieve wide instantaneous bandwidth, sophisticated high sampling rate A/D converters are required which increase system cost. In many receivers, the information bandwidth of the signals to be processed is much narrower than the instantaneous bandwidth of the receiver, allowing the sample rate to be lowered without violating the Nyquist criteria. The penalty for undersampling (utilizing a sampling rate lower than the Nyquist rate with respect to the receiver instantaneous bandwidth) is aliasing which results in frequency ambiguities. It follows that if signal processing can be provided after sampling to resolve the ambiguities to determine the true carrier frequency of the input signal, then, provided the response bandwidth of the A/D converter is greater than the receiver instantaneous bandwidth, there is no need to sample at twice the receiver instantaneous bandwidth in order to preserve signal integrity. Such processing would permit sampling at much lower rates, thereby making very efficient use of A/D converter bandwidths and materially reducing costs. The frequency ambiguities are caused by the N:1 mapping (band folding) of the receiver instantaneous bandwidth into the sample bandwidth of zero to F.sub.sample /2 for real sampling and zero to F.sub.sample for complex sampling.
It is to be understood that the discussion herein relates primarily to a basic digital receiver architecture as illustrated in FIG. 1 wherein the final intermediate frequency (IF) is discretely sampled (either real or complex) and then digitally processed. This architecture is fundamental to all digital receivers. As can be seen in FIG. 1, the analog RF input is fed to a local oscillator to derive the IF signal, this IF signal being filtered by a filter which limits the bandwidth of the incoming signal, the signal then being converted to a digital signal in an A/D converter having a sampling frequency Fs with the output of the converter sent to a processor for operation thereon. The determination of the signal frequency is a fundamental parameter to be extracted from the RF input signal.
In many digital receivers, the low pass filter preceding the A/D converter limits the signal bandwidth to one half the sample frequency (F.sub.S /2) or less. This eliminates any aliased signals but also limits the receiver bandwidth to no more than F.sub.S /2 in the case of real sampling and F.sub.S for complex sampling. Band-folded digital receivers open this bandwidth up and use signal processing techniques to estimate the true frequency of the aliased signals. Band-folded digital receivers take advantage of the fact that the input response bandwidth of many A/D converters is many times the sample bandwidth and, therefore, the corner frequency of the low pass filter preceding the A/D converter can be increased up to the response bandwidth of the A/D converter rather than be limited to the sampling frequency (F.sub.S). The A/D converter response bandwidth is a function of the input sample and hold circuit therein. As an option, an external sample and hold component can be provided t:o further increase the response bandwidth of a given A/D converter.
James B. Tsui and Richard B. Sanderson have discussed, for the real signal sampling case, a method to resolve the frequency ambiguity by sampling a delayed version of the input signal with a second A/D converter (see U.S. Pat. Nos. 5,099,194, 5,099,243, 5,109,188 and 5,235,287). Tsui and Sanderson document that the aliased frequency (F.sub.meas) and relative phase of each A/D channel can be measured using a peak detected Fast Fourier Transform (FFT) and the true frequency (F.sub.est) can be estimated from: ##EQU1## where .o slashed. is the measured phase difference between channels and T is the signal delay. This technique requires that the phase shift on the delayed signal be less than .pi. radians. This estimate is close enough to determine the alias band (n) which can be used to estimate the true frequency (F) from: EQU F=F.sub.meas +n.multidot.F.sub.S.
Tsui and Sanderson point out that this technique does not work if the input frequency is too close to an alias border or if two signals are degenerate in the lowest alias. These cases are handled by adding a second pair of A/D converters and sampling the data at a different frequency which shifts the input signals away from alias borders and resolves the degenerate case.
James B. Tsui and David L. Sharpin disclose in U.S. Pat. No. 5,198,748 a variation to the above described earlier work by eliminating the second sampling frequency and generating a second pair of IF input signals shifted in frequency by F.sub.S /4 from the first pair. Tsui and Sharpin also recognized that the signal delay can be introduced on the sample clock rather than the data, this offering a somewhat simplified architecture. Both techniques described by Tsui and Sanderson require a fixed signal delay and a d.o slashed./dt calculation as the prime discriminant for ambiguity resolution. That method requires both a frequency and a phase measurement for ambiguity resolution, whereas the invention herein requires only a frequency measurement, offering potential savings in overall system complexity.